1 /* 2 * alc5623.c -- alc562[123] ALSA Soc Audio driver 3 * 4 * Copyright 2008 Realtek Microelectronics 5 * Author: flove <flove@realtek.com> Ethan <eku@marvell.com> 6 * 7 * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org> 8 * 9 * 10 * Based on WM8753.c 11 * 12 * This program is free software; you can redistribute it and/or modify 13 * it under the terms of the GNU General Public License version 2 as 14 * published by the Free Software Foundation. 15 * 16 */ 17 18 #include <linux/module.h> 19 #include <linux/kernel.h> 20 #include <linux/init.h> 21 #include <linux/delay.h> 22 #include <linux/pm.h> 23 #include <linux/i2c.h> 24 #include <linux/regmap.h> 25 #include <linux/slab.h> 26 #include <sound/core.h> 27 #include <sound/pcm.h> 28 #include <sound/pcm_params.h> 29 #include <sound/tlv.h> 30 #include <sound/soc.h> 31 #include <sound/initval.h> 32 #include <sound/alc5623.h> 33 34 #include "alc5623.h" 35 36 static int caps_charge = 2000; 37 module_param(caps_charge, int, 0); 38 MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); 39 40 /* codec private data */ 41 struct alc5623_priv { 42 struct regmap *regmap; 43 u8 id; 44 unsigned int sysclk; 45 unsigned int add_ctrl; 46 unsigned int jack_det_ctrl; 47 }; 48 49 static inline int alc5623_reset(struct snd_soc_codec *codec) 50 { 51 return snd_soc_write(codec, ALC5623_RESET, 0); 52 } 53 54 static int amp_mixer_event(struct snd_soc_dapm_widget *w, 55 struct snd_kcontrol *kcontrol, int event) 56 { 57 /* to power-on/off class-d amp generators/speaker */ 58 /* need to write to 'index-46h' register : */ 59 /* so write index num (here 0x46) to reg 0x6a */ 60 /* and then 0xffff/0 to reg 0x6c */ 61 snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46); 62 63 switch (event) { 64 case SND_SOC_DAPM_PRE_PMU: 65 snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF); 66 break; 67 case SND_SOC_DAPM_POST_PMD: 68 snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0); 69 break; 70 } 71 72 return 0; 73 } 74 75 /* 76 * ALC5623 Controls 77 */ 78 79 static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0); 80 static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); 81 static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); 82 static const unsigned int boost_tlv[] = { 83 TLV_DB_RANGE_HEAD(3), 84 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 85 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), 86 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), 87 }; 88 static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); 89 90 static const struct snd_kcontrol_new alc5621_vol_snd_controls[] = { 91 SOC_DOUBLE_TLV("Speaker Playback Volume", 92 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), 93 SOC_DOUBLE("Speaker Playback Switch", 94 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), 95 SOC_DOUBLE_TLV("Headphone Playback Volume", 96 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), 97 SOC_DOUBLE("Headphone Playback Switch", 98 ALC5623_HP_OUT_VOL, 15, 7, 1, 1), 99 }; 100 101 static const struct snd_kcontrol_new alc5622_vol_snd_controls[] = { 102 SOC_DOUBLE_TLV("Speaker Playback Volume", 103 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), 104 SOC_DOUBLE("Speaker Playback Switch", 105 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), 106 SOC_DOUBLE_TLV("Line Playback Volume", 107 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), 108 SOC_DOUBLE("Line Playback Switch", 109 ALC5623_HP_OUT_VOL, 15, 7, 1, 1), 110 }; 111 112 static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = { 113 SOC_DOUBLE_TLV("Line Playback Volume", 114 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), 115 SOC_DOUBLE("Line Playback Switch", 116 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), 117 SOC_DOUBLE_TLV("Headphone Playback Volume", 118 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), 119 SOC_DOUBLE("Headphone Playback Switch", 120 ALC5623_HP_OUT_VOL, 15, 7, 1, 1), 121 }; 122 123 static const struct snd_kcontrol_new alc5623_snd_controls[] = { 124 SOC_DOUBLE_TLV("Auxout Playback Volume", 125 ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv), 126 SOC_DOUBLE("Auxout Playback Switch", 127 ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1), 128 SOC_DOUBLE_TLV("PCM Playback Volume", 129 ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv), 130 SOC_DOUBLE_TLV("AuxI Capture Volume", 131 ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv), 132 SOC_DOUBLE_TLV("LineIn Capture Volume", 133 ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), 134 SOC_SINGLE_TLV("Mic1 Capture Volume", 135 ALC5623_MIC_VOL, 8, 31, 1, vol_tlv), 136 SOC_SINGLE_TLV("Mic2 Capture Volume", 137 ALC5623_MIC_VOL, 0, 31, 1, vol_tlv), 138 SOC_DOUBLE_TLV("Rec Capture Volume", 139 ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv), 140 SOC_SINGLE_TLV("Mic 1 Boost Volume", 141 ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv), 142 SOC_SINGLE_TLV("Mic 2 Boost Volume", 143 ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv), 144 SOC_SINGLE_TLV("Digital Boost Volume", 145 ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv), 146 }; 147 148 /* 149 * DAPM Controls 150 */ 151 static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = { 152 SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1), 153 SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1), 154 SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1), 155 SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1), 156 SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1), 157 }; 158 159 static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = { 160 SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1), 161 }; 162 163 static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = { 164 SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1), 165 }; 166 167 static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = { 168 SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1), 169 SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1), 170 SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1), 171 SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1), 172 SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1), 173 SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1), 174 SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1), 175 }; 176 177 static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = { 178 SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1), 179 SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1), 180 SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1), 181 SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1), 182 SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1), 183 }; 184 185 /* Left Record Mixer */ 186 static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = { 187 SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1), 188 SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1), 189 SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1), 190 SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1), 191 SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1), 192 SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1), 193 SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1), 194 }; 195 196 /* Right Record Mixer */ 197 static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = { 198 SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1), 199 SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1), 200 SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1), 201 SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1), 202 SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1), 203 SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1), 204 SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1), 205 }; 206 207 static const char *alc5623_spk_n_sour_sel[] = { 208 "RN/-R", "RP/+R", "LN/-R", "Vmid" }; 209 static const char *alc5623_hpl_out_input_sel[] = { 210 "Vmid", "HP Left Mix"}; 211 static const char *alc5623_hpr_out_input_sel[] = { 212 "Vmid", "HP Right Mix"}; 213 static const char *alc5623_spkout_input_sel[] = { 214 "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; 215 static const char *alc5623_aux_out_input_sel[] = { 216 "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; 217 218 /* auxout output mux */ 219 static SOC_ENUM_SINGLE_DECL(alc5623_aux_out_input_enum, 220 ALC5623_OUTPUT_MIXER_CTRL, 6, 221 alc5623_aux_out_input_sel); 222 static const struct snd_kcontrol_new alc5623_auxout_mux_controls = 223 SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum); 224 225 /* speaker output mux */ 226 static SOC_ENUM_SINGLE_DECL(alc5623_spkout_input_enum, 227 ALC5623_OUTPUT_MIXER_CTRL, 10, 228 alc5623_spkout_input_sel); 229 static const struct snd_kcontrol_new alc5623_spkout_mux_controls = 230 SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum); 231 232 /* headphone left output mux */ 233 static SOC_ENUM_SINGLE_DECL(alc5623_hpl_out_input_enum, 234 ALC5623_OUTPUT_MIXER_CTRL, 9, 235 alc5623_hpl_out_input_sel); 236 static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls = 237 SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum); 238 239 /* headphone right output mux */ 240 static SOC_ENUM_SINGLE_DECL(alc5623_hpr_out_input_enum, 241 ALC5623_OUTPUT_MIXER_CTRL, 8, 242 alc5623_hpr_out_input_sel); 243 static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls = 244 SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum); 245 246 /* speaker output N select */ 247 static SOC_ENUM_SINGLE_DECL(alc5623_spk_n_sour_enum, 248 ALC5623_OUTPUT_MIXER_CTRL, 14, 249 alc5623_spk_n_sour_sel); 250 static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls = 251 SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum); 252 253 static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = { 254 /* Muxes */ 255 SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0, 256 &alc5623_auxout_mux_controls), 257 SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0, 258 &alc5623_spkout_mux_controls), 259 SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, 260 &alc5623_hpl_out_mux_controls), 261 SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, 262 &alc5623_hpr_out_mux_controls), 263 SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, 264 &alc5623_spkoutn_mux_controls), 265 266 /* output mixers */ 267 SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, 268 &alc5623_hp_mixer_controls[0], 269 ARRAY_SIZE(alc5623_hp_mixer_controls)), 270 SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0, 271 &alc5623_hpr_mixer_controls[0], 272 ARRAY_SIZE(alc5623_hpr_mixer_controls)), 273 SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0, 274 &alc5623_hpl_mixer_controls[0], 275 ARRAY_SIZE(alc5623_hpl_mixer_controls)), 276 SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0), 277 SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0, 278 &alc5623_mono_mixer_controls[0], 279 ARRAY_SIZE(alc5623_mono_mixer_controls)), 280 SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0, 281 &alc5623_speaker_mixer_controls[0], 282 ARRAY_SIZE(alc5623_speaker_mixer_controls)), 283 284 /* input mixers */ 285 SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0, 286 &alc5623_captureL_mixer_controls[0], 287 ARRAY_SIZE(alc5623_captureL_mixer_controls)), 288 SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0, 289 &alc5623_captureR_mixer_controls[0], 290 ARRAY_SIZE(alc5623_captureR_mixer_controls)), 291 292 SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", 293 ALC5623_PWR_MANAG_ADD2, 9, 0), 294 SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", 295 ALC5623_PWR_MANAG_ADD2, 8, 0), 296 SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0), 297 SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0), 298 SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), 299 SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", 300 ALC5623_PWR_MANAG_ADD2, 7, 0), 301 SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", 302 ALC5623_PWR_MANAG_ADD2, 6, 0), 303 SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0), 304 SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0), 305 SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0), 306 SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0), 307 SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0), 308 SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0), 309 SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0), 310 SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0), 311 SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0), 312 SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0), 313 SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0), 314 SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0), 315 SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0), 316 SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0), 317 318 SND_SOC_DAPM_OUTPUT("AUXOUTL"), 319 SND_SOC_DAPM_OUTPUT("AUXOUTR"), 320 SND_SOC_DAPM_OUTPUT("HPL"), 321 SND_SOC_DAPM_OUTPUT("HPR"), 322 SND_SOC_DAPM_OUTPUT("SPKOUT"), 323 SND_SOC_DAPM_OUTPUT("SPKOUTN"), 324 SND_SOC_DAPM_INPUT("LINEINL"), 325 SND_SOC_DAPM_INPUT("LINEINR"), 326 SND_SOC_DAPM_INPUT("AUXINL"), 327 SND_SOC_DAPM_INPUT("AUXINR"), 328 SND_SOC_DAPM_INPUT("MIC1"), 329 SND_SOC_DAPM_INPUT("MIC2"), 330 SND_SOC_DAPM_VMID("Vmid"), 331 }; 332 333 static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"}; 334 static SOC_ENUM_SINGLE_DECL(alc5623_amp_enum, 335 ALC5623_OUTPUT_MIXER_CTRL, 13, 336 alc5623_amp_names); 337 static const struct snd_kcontrol_new alc5623_amp_mux_controls = 338 SOC_DAPM_ENUM("Route", alc5623_amp_enum); 339 340 static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = { 341 SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0, 342 amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), 343 SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0), 344 SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0, 345 &alc5623_amp_mux_controls), 346 }; 347 348 static const struct snd_soc_dapm_route intercon[] = { 349 /* virtual mixer - mixes left & right channels */ 350 {"I2S Mix", NULL, "Left DAC"}, 351 {"I2S Mix", NULL, "Right DAC"}, 352 {"Line Mix", NULL, "Right LineIn"}, 353 {"Line Mix", NULL, "Left LineIn"}, 354 {"AuxI Mix", NULL, "Left AuxI"}, 355 {"AuxI Mix", NULL, "Right AuxI"}, 356 {"AUXOUTL", NULL, "Left AuxOut"}, 357 {"AUXOUTR", NULL, "Right AuxOut"}, 358 359 /* HP mixer */ 360 {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"}, 361 {"HPL Mix", NULL, "HP Mix"}, 362 {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"}, 363 {"HPR Mix", NULL, "HP Mix"}, 364 {"HP Mix", "LI2HP Playback Switch", "Line Mix"}, 365 {"HP Mix", "AUXI2HP Playback Switch", "AuxI Mix"}, 366 {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, 367 {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, 368 {"HP Mix", "DAC2HP Playback Switch", "I2S Mix"}, 369 370 /* speaker mixer */ 371 {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, 372 {"Speaker Mix", "AUXI2SPK Playback Switch", "AuxI Mix"}, 373 {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, 374 {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, 375 {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"}, 376 377 /* mono mixer */ 378 {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, 379 {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, 380 {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, 381 {"Mono Mix", "AUXI2MONO Playback Switch", "AuxI Mix"}, 382 {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, 383 {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, 384 {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"}, 385 386 /* Left record mixer */ 387 {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, 388 {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"}, 389 {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, 390 {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, 391 {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, 392 {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, 393 {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, 394 395 /*Right record mixer */ 396 {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, 397 {"Right Capture Mix", "Right AuxI Capture Switch", "AUXINR"}, 398 {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, 399 {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, 400 {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, 401 {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, 402 {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, 403 404 /* headphone left mux */ 405 {"Left Headphone Mux", "HP Left Mix", "HPL Mix"}, 406 {"Left Headphone Mux", "Vmid", "Vmid"}, 407 408 /* headphone right mux */ 409 {"Right Headphone Mux", "HP Right Mix", "HPR Mix"}, 410 {"Right Headphone Mux", "Vmid", "Vmid"}, 411 412 /* speaker out mux */ 413 {"SpeakerOut Mux", "Vmid", "Vmid"}, 414 {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"}, 415 {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"}, 416 {"SpeakerOut Mux", "Mono Mix", "Mono Mix"}, 417 418 /* Mono/Aux Out mux */ 419 {"AuxOut Mux", "Vmid", "Vmid"}, 420 {"AuxOut Mux", "HPOut Mix", "HPOut Mix"}, 421 {"AuxOut Mux", "Speaker Mix", "Speaker Mix"}, 422 {"AuxOut Mux", "Mono Mix", "Mono Mix"}, 423 424 /* output pga */ 425 {"HPL", NULL, "Left Headphone"}, 426 {"Left Headphone", NULL, "Left Headphone Mux"}, 427 {"HPR", NULL, "Right Headphone"}, 428 {"Right Headphone", NULL, "Right Headphone Mux"}, 429 {"Left AuxOut", NULL, "AuxOut Mux"}, 430 {"Right AuxOut", NULL, "AuxOut Mux"}, 431 432 /* input pga */ 433 {"Left LineIn", NULL, "LINEINL"}, 434 {"Right LineIn", NULL, "LINEINR"}, 435 {"Left AuxI", NULL, "AUXINL"}, 436 {"Right AuxI", NULL, "AUXINR"}, 437 {"MIC1 Pre Amp", NULL, "MIC1"}, 438 {"MIC2 Pre Amp", NULL, "MIC2"}, 439 {"MIC1 PGA", NULL, "MIC1 Pre Amp"}, 440 {"MIC2 PGA", NULL, "MIC2 Pre Amp"}, 441 442 /* left ADC */ 443 {"Left ADC", NULL, "Left Capture Mix"}, 444 445 /* right ADC */ 446 {"Right ADC", NULL, "Right Capture Mix"}, 447 448 {"SpeakerOut N Mux", "RN/-R", "SpeakerOut"}, 449 {"SpeakerOut N Mux", "RP/+R", "SpeakerOut"}, 450 {"SpeakerOut N Mux", "LN/-R", "SpeakerOut"}, 451 {"SpeakerOut N Mux", "Vmid", "Vmid"}, 452 453 {"SPKOUT", NULL, "SpeakerOut"}, 454 {"SPKOUTN", NULL, "SpeakerOut N Mux"}, 455 }; 456 457 static const struct snd_soc_dapm_route intercon_spk[] = { 458 {"SpeakerOut", NULL, "SpeakerOut Mux"}, 459 }; 460 461 static const struct snd_soc_dapm_route intercon_amp_spk[] = { 462 {"AB Amp", NULL, "SpeakerOut Mux"}, 463 {"D Amp", NULL, "SpeakerOut Mux"}, 464 {"AB-D Amp Mux", "AB Amp", "AB Amp"}, 465 {"AB-D Amp Mux", "D Amp", "D Amp"}, 466 {"SpeakerOut", NULL, "AB-D Amp Mux"}, 467 }; 468 469 /* PLL divisors */ 470 struct _pll_div { 471 u32 pll_in; 472 u32 pll_out; 473 u16 regvalue; 474 }; 475 476 /* Note : pll code from original alc5623 driver. Not sure of how good it is */ 477 /* useful only for master mode */ 478 static const struct _pll_div codec_master_pll_div[] = { 479 480 { 2048000, 8192000, 0x0ea0}, 481 { 3686400, 8192000, 0x4e27}, 482 { 12000000, 8192000, 0x456b}, 483 { 13000000, 8192000, 0x495f}, 484 { 13100000, 8192000, 0x0320}, 485 { 2048000, 11289600, 0xf637}, 486 { 3686400, 11289600, 0x2f22}, 487 { 12000000, 11289600, 0x3e2f}, 488 { 13000000, 11289600, 0x4d5b}, 489 { 13100000, 11289600, 0x363b}, 490 { 2048000, 16384000, 0x1ea0}, 491 { 3686400, 16384000, 0x9e27}, 492 { 12000000, 16384000, 0x452b}, 493 { 13000000, 16384000, 0x542f}, 494 { 13100000, 16384000, 0x03a0}, 495 { 2048000, 16934400, 0xe625}, 496 { 3686400, 16934400, 0x9126}, 497 { 12000000, 16934400, 0x4d2c}, 498 { 13000000, 16934400, 0x742f}, 499 { 13100000, 16934400, 0x3c27}, 500 { 2048000, 22579200, 0x2aa0}, 501 { 3686400, 22579200, 0x2f20}, 502 { 12000000, 22579200, 0x7e2f}, 503 { 13000000, 22579200, 0x742f}, 504 { 13100000, 22579200, 0x3c27}, 505 { 2048000, 24576000, 0x2ea0}, 506 { 3686400, 24576000, 0xee27}, 507 { 12000000, 24576000, 0x2915}, 508 { 13000000, 24576000, 0x772e}, 509 { 13100000, 24576000, 0x0d20}, 510 }; 511 512 static const struct _pll_div codec_slave_pll_div[] = { 513 514 { 1024000, 16384000, 0x3ea0}, 515 { 1411200, 22579200, 0x3ea0}, 516 { 1536000, 24576000, 0x3ea0}, 517 { 2048000, 16384000, 0x1ea0}, 518 { 2822400, 22579200, 0x1ea0}, 519 { 3072000, 24576000, 0x1ea0}, 520 521 }; 522 523 static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, 524 int source, unsigned int freq_in, unsigned int freq_out) 525 { 526 int i; 527 struct snd_soc_codec *codec = codec_dai->codec; 528 int gbl_clk = 0, pll_div = 0; 529 u16 reg; 530 531 if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK) 532 return -ENODEV; 533 534 /* Disable PLL power */ 535 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2, 536 ALC5623_PWR_ADD2_PLL, 537 0); 538 539 /* pll is not used in slave mode */ 540 reg = snd_soc_read(codec, ALC5623_DAI_CONTROL); 541 if (reg & ALC5623_DAI_SDP_SLAVE_MODE) 542 return 0; 543 544 if (!freq_in || !freq_out) 545 return 0; 546 547 switch (pll_id) { 548 case ALC5623_PLL_FR_MCLK: 549 for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) { 550 if (codec_master_pll_div[i].pll_in == freq_in 551 && codec_master_pll_div[i].pll_out == freq_out) { 552 /* PLL source from MCLK */ 553 pll_div = codec_master_pll_div[i].regvalue; 554 break; 555 } 556 } 557 break; 558 case ALC5623_PLL_FR_BCK: 559 for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { 560 if (codec_slave_pll_div[i].pll_in == freq_in 561 && codec_slave_pll_div[i].pll_out == freq_out) { 562 /* PLL source from Bitclk */ 563 gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK; 564 pll_div = codec_slave_pll_div[i].regvalue; 565 break; 566 } 567 } 568 break; 569 default: 570 return -EINVAL; 571 } 572 573 if (!pll_div) 574 return -EINVAL; 575 576 snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk); 577 snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div); 578 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2, 579 ALC5623_PWR_ADD2_PLL, 580 ALC5623_PWR_ADD2_PLL); 581 gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL; 582 snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk); 583 584 return 0; 585 } 586 587 struct _coeff_div { 588 u16 fs; 589 u16 regvalue; 590 }; 591 592 /* codec hifi mclk (after PLL) clock divider coefficients */ 593 /* values inspired from column BCLK=32Fs of Appendix A table */ 594 static const struct _coeff_div coeff_div[] = { 595 {256*8, 0x3a69}, 596 {384*8, 0x3c6b}, 597 {256*4, 0x2a69}, 598 {384*4, 0x2c6b}, 599 {256*2, 0x1a69}, 600 {384*2, 0x1c6b}, 601 {256*1, 0x0a69}, 602 {384*1, 0x0c6b}, 603 }; 604 605 static int get_coeff(struct snd_soc_codec *codec, int rate) 606 { 607 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); 608 int i; 609 610 for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { 611 if (coeff_div[i].fs * rate == alc5623->sysclk) 612 return i; 613 } 614 return -EINVAL; 615 } 616 617 /* 618 * Clock after PLL and dividers 619 */ 620 static int alc5623_set_dai_sysclk(struct snd_soc_dai *codec_dai, 621 int clk_id, unsigned int freq, int dir) 622 { 623 struct snd_soc_codec *codec = codec_dai->codec; 624 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); 625 626 switch (freq) { 627 case 8192000: 628 case 11289600: 629 case 12288000: 630 case 16384000: 631 case 16934400: 632 case 18432000: 633 case 22579200: 634 case 24576000: 635 alc5623->sysclk = freq; 636 return 0; 637 } 638 return -EINVAL; 639 } 640 641 static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, 642 unsigned int fmt) 643 { 644 struct snd_soc_codec *codec = codec_dai->codec; 645 u16 iface = 0; 646 647 /* set master/slave audio interface */ 648 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { 649 case SND_SOC_DAIFMT_CBM_CFM: 650 iface = ALC5623_DAI_SDP_MASTER_MODE; 651 break; 652 case SND_SOC_DAIFMT_CBS_CFS: 653 iface = ALC5623_DAI_SDP_SLAVE_MODE; 654 break; 655 default: 656 return -EINVAL; 657 } 658 659 /* interface format */ 660 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { 661 case SND_SOC_DAIFMT_I2S: 662 iface |= ALC5623_DAI_I2S_DF_I2S; 663 break; 664 case SND_SOC_DAIFMT_RIGHT_J: 665 iface |= ALC5623_DAI_I2S_DF_RIGHT; 666 break; 667 case SND_SOC_DAIFMT_LEFT_J: 668 iface |= ALC5623_DAI_I2S_DF_LEFT; 669 break; 670 case SND_SOC_DAIFMT_DSP_A: 671 iface |= ALC5623_DAI_I2S_DF_PCM; 672 break; 673 case SND_SOC_DAIFMT_DSP_B: 674 iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE; 675 break; 676 default: 677 return -EINVAL; 678 } 679 680 /* clock inversion */ 681 switch (fmt & SND_SOC_DAIFMT_INV_MASK) { 682 case SND_SOC_DAIFMT_NB_NF: 683 break; 684 case SND_SOC_DAIFMT_IB_IF: 685 iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL; 686 break; 687 case SND_SOC_DAIFMT_IB_NF: 688 iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL; 689 break; 690 case SND_SOC_DAIFMT_NB_IF: 691 break; 692 default: 693 return -EINVAL; 694 } 695 696 return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface); 697 } 698 699 static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream, 700 struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) 701 { 702 struct snd_soc_codec *codec = dai->codec; 703 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); 704 int coeff, rate; 705 u16 iface; 706 707 iface = snd_soc_read(codec, ALC5623_DAI_CONTROL); 708 iface &= ~ALC5623_DAI_I2S_DL_MASK; 709 710 /* bit size */ 711 switch (params_width(params)) { 712 case 16: 713 iface |= ALC5623_DAI_I2S_DL_16; 714 break; 715 case 20: 716 iface |= ALC5623_DAI_I2S_DL_20; 717 break; 718 case 24: 719 iface |= ALC5623_DAI_I2S_DL_24; 720 break; 721 case 32: 722 iface |= ALC5623_DAI_I2S_DL_32; 723 break; 724 default: 725 return -EINVAL; 726 } 727 728 /* set iface & srate */ 729 snd_soc_write(codec, ALC5623_DAI_CONTROL, iface); 730 rate = params_rate(params); 731 coeff = get_coeff(codec, rate); 732 if (coeff < 0) 733 return -EINVAL; 734 735 coeff = coeff_div[coeff].regvalue; 736 dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n", 737 __func__, alc5623->sysclk, rate, coeff); 738 snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff); 739 740 return 0; 741 } 742 743 static int alc5623_mute(struct snd_soc_dai *dai, int mute) 744 { 745 struct snd_soc_codec *codec = dai->codec; 746 u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT; 747 u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute; 748 749 if (mute) 750 mute_reg |= hp_mute; 751 752 return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg); 753 } 754 755 #define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \ 756 | ALC5623_PWR_ADD2_DAC_REF_CIR) 757 758 #define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \ 759 | ALC5623_PWR_ADD3_MIC1_BOOST_AD) 760 761 #define ALC5623_ADD1_POWER_EN \ 762 (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \ 763 | ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \ 764 | ALC5623_PWR_ADD1_HP_OUT_ENH_AMP) 765 766 #define ALC5623_ADD1_POWER_EN_5622 \ 767 (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \ 768 | ALC5623_PWR_ADD1_HP_OUT_AMP) 769 770 static void enable_power_depop(struct snd_soc_codec *codec) 771 { 772 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); 773 774 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1, 775 ALC5623_PWR_ADD1_SOFTGEN_EN, 776 ALC5623_PWR_ADD1_SOFTGEN_EN); 777 778 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN); 779 780 snd_soc_update_bits(codec, ALC5623_MISC_CTRL, 781 ALC5623_MISC_HP_DEPOP_MODE2_EN, 782 ALC5623_MISC_HP_DEPOP_MODE2_EN); 783 784 msleep(500); 785 786 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN); 787 788 /* avoid writing '1' into 5622 reserved bits */ 789 if (alc5623->id == 0x22) 790 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 791 ALC5623_ADD1_POWER_EN_5622); 792 else 793 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 794 ALC5623_ADD1_POWER_EN); 795 796 /* disable HP Depop2 */ 797 snd_soc_update_bits(codec, ALC5623_MISC_CTRL, 798 ALC5623_MISC_HP_DEPOP_MODE2_EN, 799 0); 800 801 } 802 803 static int alc5623_set_bias_level(struct snd_soc_codec *codec, 804 enum snd_soc_bias_level level) 805 { 806 switch (level) { 807 case SND_SOC_BIAS_ON: 808 enable_power_depop(codec); 809 break; 810 case SND_SOC_BIAS_PREPARE: 811 break; 812 case SND_SOC_BIAS_STANDBY: 813 /* everything off except vref/vmid, */ 814 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 815 ALC5623_PWR_ADD2_VREF); 816 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 817 ALC5623_PWR_ADD3_MAIN_BIAS); 818 break; 819 case SND_SOC_BIAS_OFF: 820 /* everything off, dac mute, inactive */ 821 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0); 822 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0); 823 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0); 824 break; 825 } 826 codec->dapm.bias_level = level; 827 return 0; 828 } 829 830 #define ALC5623_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \ 831 | SNDRV_PCM_FMTBIT_S24_LE \ 832 | SNDRV_PCM_FMTBIT_S32_LE) 833 834 static const struct snd_soc_dai_ops alc5623_dai_ops = { 835 .hw_params = alc5623_pcm_hw_params, 836 .digital_mute = alc5623_mute, 837 .set_fmt = alc5623_set_dai_fmt, 838 .set_sysclk = alc5623_set_dai_sysclk, 839 .set_pll = alc5623_set_dai_pll, 840 }; 841 842 static struct snd_soc_dai_driver alc5623_dai = { 843 .name = "alc5623-hifi", 844 .playback = { 845 .stream_name = "Playback", 846 .channels_min = 1, 847 .channels_max = 2, 848 .rate_min = 8000, 849 .rate_max = 48000, 850 .rates = SNDRV_PCM_RATE_8000_48000, 851 .formats = ALC5623_FORMATS,}, 852 .capture = { 853 .stream_name = "Capture", 854 .channels_min = 1, 855 .channels_max = 2, 856 .rate_min = 8000, 857 .rate_max = 48000, 858 .rates = SNDRV_PCM_RATE_8000_48000, 859 .formats = ALC5623_FORMATS,}, 860 861 .ops = &alc5623_dai_ops, 862 }; 863 864 static int alc5623_suspend(struct snd_soc_codec *codec) 865 { 866 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); 867 868 alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); 869 regcache_cache_only(alc5623->regmap, true); 870 871 return 0; 872 } 873 874 static int alc5623_resume(struct snd_soc_codec *codec) 875 { 876 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); 877 int ret; 878 879 /* Sync reg_cache with the hardware */ 880 regcache_cache_only(alc5623->regmap, false); 881 ret = regcache_sync(alc5623->regmap); 882 if (ret != 0) { 883 dev_err(codec->dev, "Failed to sync register cache: %d\n", 884 ret); 885 regcache_cache_only(alc5623->regmap, true); 886 return ret; 887 } 888 889 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); 890 891 /* charge alc5623 caps */ 892 if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { 893 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); 894 codec->dapm.bias_level = SND_SOC_BIAS_ON; 895 alc5623_set_bias_level(codec, codec->dapm.bias_level); 896 } 897 898 return 0; 899 } 900 901 static int alc5623_probe(struct snd_soc_codec *codec) 902 { 903 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); 904 struct snd_soc_dapm_context *dapm = &codec->dapm; 905 906 alc5623_reset(codec); 907 908 /* power on device */ 909 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); 910 911 if (alc5623->add_ctrl) { 912 snd_soc_write(codec, ALC5623_ADD_CTRL_REG, 913 alc5623->add_ctrl); 914 } 915 916 if (alc5623->jack_det_ctrl) { 917 snd_soc_write(codec, ALC5623_JACK_DET_CTRL, 918 alc5623->jack_det_ctrl); 919 } 920 921 switch (alc5623->id) { 922 case 0x21: 923 snd_soc_add_codec_controls(codec, alc5621_vol_snd_controls, 924 ARRAY_SIZE(alc5621_vol_snd_controls)); 925 break; 926 case 0x22: 927 snd_soc_add_codec_controls(codec, alc5622_vol_snd_controls, 928 ARRAY_SIZE(alc5622_vol_snd_controls)); 929 break; 930 case 0x23: 931 snd_soc_add_codec_controls(codec, alc5623_vol_snd_controls, 932 ARRAY_SIZE(alc5623_vol_snd_controls)); 933 break; 934 default: 935 return -EINVAL; 936 } 937 938 snd_soc_add_codec_controls(codec, alc5623_snd_controls, 939 ARRAY_SIZE(alc5623_snd_controls)); 940 941 snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets, 942 ARRAY_SIZE(alc5623_dapm_widgets)); 943 944 /* set up audio path interconnects */ 945 snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); 946 947 switch (alc5623->id) { 948 case 0x21: 949 case 0x22: 950 snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets, 951 ARRAY_SIZE(alc5623_dapm_amp_widgets)); 952 snd_soc_dapm_add_routes(dapm, intercon_amp_spk, 953 ARRAY_SIZE(intercon_amp_spk)); 954 break; 955 case 0x23: 956 snd_soc_dapm_add_routes(dapm, intercon_spk, 957 ARRAY_SIZE(intercon_spk)); 958 break; 959 default: 960 return -EINVAL; 961 } 962 963 return 0; 964 } 965 966 /* power down chip */ 967 static int alc5623_remove(struct snd_soc_codec *codec) 968 { 969 alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); 970 return 0; 971 } 972 973 static struct snd_soc_codec_driver soc_codec_device_alc5623 = { 974 .probe = alc5623_probe, 975 .remove = alc5623_remove, 976 .suspend = alc5623_suspend, 977 .resume = alc5623_resume, 978 .set_bias_level = alc5623_set_bias_level, 979 }; 980 981 static const struct regmap_config alc5623_regmap = { 982 .reg_bits = 8, 983 .val_bits = 16, 984 .reg_stride = 2, 985 986 .max_register = ALC5623_VENDOR_ID2, 987 .cache_type = REGCACHE_RBTREE, 988 }; 989 990 /* 991 * ALC5623 2 wire address is determined by A1 pin 992 * state during powerup. 993 * low = 0x1a 994 * high = 0x1b 995 */ 996 static int alc5623_i2c_probe(struct i2c_client *client, 997 const struct i2c_device_id *id) 998 { 999 struct alc5623_platform_data *pdata; 1000 struct alc5623_priv *alc5623; 1001 unsigned int vid1, vid2; 1002 int ret; 1003 1004 alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), 1005 GFP_KERNEL); 1006 if (alc5623 == NULL) 1007 return -ENOMEM; 1008 1009 alc5623->regmap = devm_regmap_init_i2c(client, &alc5623_regmap); 1010 if (IS_ERR(alc5623->regmap)) { 1011 ret = PTR_ERR(alc5623->regmap); 1012 dev_err(&client->dev, "Failed to initialise I/O: %d\n", ret); 1013 return ret; 1014 } 1015 1016 ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID1, &vid1); 1017 if (ret < 0) { 1018 dev_err(&client->dev, "failed to read vendor ID1: %d\n", ret); 1019 return ret; 1020 } 1021 vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); 1022 1023 ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2); 1024 if (ret < 0) { 1025 dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret); 1026 return ret; 1027 } 1028 1029 if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { 1030 dev_err(&client->dev, "unknown or wrong codec\n"); 1031 dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n", 1032 0x10ec, id->driver_data, 1033 vid1, vid2); 1034 return -ENODEV; 1035 } 1036 1037 dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2); 1038 1039 pdata = client->dev.platform_data; 1040 if (pdata) { 1041 alc5623->add_ctrl = pdata->add_ctrl; 1042 alc5623->jack_det_ctrl = pdata->jack_det_ctrl; 1043 } 1044 1045 alc5623->id = vid2; 1046 switch (alc5623->id) { 1047 case 0x21: 1048 alc5623_dai.name = "alc5621-hifi"; 1049 break; 1050 case 0x22: 1051 alc5623_dai.name = "alc5622-hifi"; 1052 break; 1053 case 0x23: 1054 alc5623_dai.name = "alc5623-hifi"; 1055 break; 1056 default: 1057 return -EINVAL; 1058 } 1059 1060 i2c_set_clientdata(client, alc5623); 1061 1062 ret = snd_soc_register_codec(&client->dev, 1063 &soc_codec_device_alc5623, &alc5623_dai, 1); 1064 if (ret != 0) 1065 dev_err(&client->dev, "Failed to register codec: %d\n", ret); 1066 1067 return ret; 1068 } 1069 1070 static int alc5623_i2c_remove(struct i2c_client *client) 1071 { 1072 snd_soc_unregister_codec(&client->dev); 1073 return 0; 1074 } 1075 1076 static const struct i2c_device_id alc5623_i2c_table[] = { 1077 {"alc5621", 0x21}, 1078 {"alc5622", 0x22}, 1079 {"alc5623", 0x23}, 1080 {} 1081 }; 1082 MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table); 1083 1084 /* i2c codec control layer */ 1085 static struct i2c_driver alc5623_i2c_driver = { 1086 .driver = { 1087 .name = "alc562x-codec", 1088 .owner = THIS_MODULE, 1089 }, 1090 .probe = alc5623_i2c_probe, 1091 .remove = alc5623_i2c_remove, 1092 .id_table = alc5623_i2c_table, 1093 }; 1094 1095 module_i2c_driver(alc5623_i2c_driver); 1096 1097 MODULE_DESCRIPTION("ASoC alc5621/2/3 driver"); 1098 MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); 1099 MODULE_LICENSE("GPL"); 1100